IRC log for #asterisk on 20190628

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07:03.02*** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.0 (2019/5/30) 16.4.0 (2019/5/30), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
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10:19.20Asteriscohi
10:19.27Asteriscosomeone could help me?
10:19.56Asteriscoi need call from a number line
10:20.13Asteriscoif this number is busy i need call with a secondary line
10:20.31Asteriscoand if i call with this line i need play an audio file
10:49.57Asteriscozzzzzzzz
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13:47.03Asteriscoit's possible concatenate two comand in extensions.conf? Dial and Playback
13:47.38SamotNo.
13:48.11SamotThat is why Dial() has pre/post dial options to send calls to contexts for that
13:48.39Asteriscoi need play a wav file before that i call with a line number
13:49.38Asteriscoif the line is busy i don't listen the playback file
13:49.42Asteriscohow can i do it?
13:51.00SamotYou can't.
13:51.19SamotYou either play the file before the lines are connected or after they are connected.
13:51.29SamotIf the destination is busy, the line will not be connected.
13:51.42Asterisco:\
13:51.53Asterisconot the destination
13:52.03Asteriscobut if is busy my phone line
13:52.11SamotWhat?
13:52.18Asteriscoi've 3 line number in outgoing
13:52.37Asteriscotwo of it have callerid visible
13:52.47Asteriscothe thirst line is anonymous
13:53.22Asteriscowhen i use the anonymous line, i need playback a file that tell me that i'm using the anonymous line
13:54.20AsteriscoSamot?
13:54.45SamotSo you have your phone...
13:54.52SamotThat _you_ are dialing from.
13:54.56SamotIt has three lines on it?
13:55.04Asteriscoyes
13:55.13SamotAnd one of those lines is the anonymous line?
13:55.17Asteriscotwo with callerid visible
13:55.22Asterisco1 anonymous line
13:55.25Asteriscoyes
13:55.39SamotOne of the line programmed on the phone is for the anonymous calls?
13:55.54Asteriscono
13:56.03SamotSo how do you get the anonymous line?
13:56.39Asterisco...in first time i try to use the first line with callerid visible
13:56.56Asteriscoif the first line is visible i need use the second line that is anonymous
13:56.57sibiriaone way you could do it is by inspecting the channel in the post-dial gosub
13:57.01SamotOK..
13:57.06Asteriscoif the second line is also busy
13:57.11SamotHow do you pick which of those three lines to use?
13:57.21SamotHow is your dialplan setup for that?
13:57.21Asteriscoi need use the thirst line that is with callerid visible
13:57.49Asteriscosimple
13:57.50Asterisco...
13:58.00Asteriscoexten => _0X.,1,Dial(SIP/${EXTEN}@095448212) ;
13:58.00sibiriaie. Dial(PJSIP/blah,,U(postdial_context))
13:58.13SamotIs that a callerID line?
13:58.13Asteriscoexten => _0X.,n,Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
13:58.20SamotOr the anonymous?
13:58.25Asteriscoexten => _0X.,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30) ;T
13:58.34Asteriscoexten => _0X.,n,Dial(SIP/${EXTEN}@0955183828) ;
13:59.00Samot<PROTECTED>
13:59.03Asteriscothe iax line is anonymous
13:59.05sibiriaAsterisco: Dial() is blocking and will not proceed until the call fails or is hung-up
13:59.19Asteriscoexten => _0X.,n,Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
13:59.24sibiriayou need to do this in a post-dial context
13:59.31sibirialike Samot explained minutes ago
13:59.38Asterisco????
13:59.41Samotsibiria: No, he wants to know before he dials.
14:00.09sibiriadidn't he just say he wants to hear a certain indication when the call is bridged on the anon peer?
14:00.18SamotAsterisco: Are all these in the same context?
14:00.24Asteriscoyes
14:00.29SamotShow the context
14:00.37Asteriscoif i use playback in this mode...
14:00.54Asterisco....i hear the advice alse if i use the thirst line
14:01.36AsteriscoSamot is [outgoing] context
14:01.54SamotI meant show what you have it in. pastebin it
14:02.48Asteriscohttps://pastebin.com/t1HYqTCW
14:03.29Asteriscoi need concatenate the playback with the IAX call
14:04.45AsteriscoSamot?
14:05.49SamotHow does it even get to it?
14:06.56Asterisco[interni]
14:07.03Asteriscoinclude => outgoing
14:07.31Samotexten => _0X.,1,Dial(SIP/${EXTEN}@095448212) <-- this is always executed first.
14:07.38*** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com)
14:07.46Asteriscoyes
14:08.22SamotSo when is exten => _0X.,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30)  supposed to be used?
14:08.43Asteriscoyes
14:08.48SamotWHEN?
14:08.48Asteriscoonly with it
14:08.55SamotHow do you use it?
14:08.57Asteriscofirst that start the call
14:09.37Asteriscothe file that i pastebine workfine...
14:09.50Asteriscobut if the first and second line are busy
14:10.06sibiriathat dial plan will try call on all three lines, no matter what
14:10.10Asteriscostart the thirst line but i hear the advice "chiamata_anonima"
14:10.36sibiriawhat is "thirst"? first (1st) or third (3rd)?
14:10.48Asteriscosibiria look the pastebin
14:10.58Asteriscohttps://pastebin.com/t1HYqTCW
14:11.21Asterisco1) 095448212
14:11.24Asterisco2) IAX
14:11.25sibiriayes i'm just inquiring about your choice of word, because thirst is the english word for sete
14:11.34Asterisco3)0955183828
14:12.07Asteriscothird
14:12.10Asteriscopardon :D
14:12.38sibiriaok anyway, that context will try make three calls
14:12.42sibiriaafter eachother
14:12.55Asteriscoin that order
14:12.56sibiriaeven if the first is answered
14:13.34Asteriscoi need playback only if IAX call is not busy
14:14.07SamotBusy how?
14:14.17SamotThe other side sends back a busy?
14:14.24sibiriawith Dial() you have the option to do something when the call is accepted but before it's bridged
14:14.35sibiriaand, also, you have the option to do something when the call is bridged
14:14.36Asteriscoif this line is used from ather people
14:14.45SamotHow does it know?
14:14.51SamotAre you checking a state somewhere?
14:15.05SamotHow does Asterisk know that the IAX  trunk is "busy"
14:15.23sibiriait doesn't, it's why i adviced him to do this in a post-dial context which inspects the channel
14:15.31AsteriscoSamot and Sibiria...
14:15.43Asterisco....my context works fine
14:16.15Asterisco....but i need adjust it
14:16.17sibiriayes we understand it
14:16.30sibiriayou want a special audio file to be played to you if the IAX channel successfully bridges the call
14:16.38Asteriscoyes :)
14:16.47Asteriscois not possible do it?
14:16.49sibiriai am now telling you for the fourth and last time: you do this using a POST-DIAL GOSUB INSIDE DIAL()
14:16.50Asterisco:(
14:17.02sibiriaPOST-DIAL GOSUB INSIDE DIAL(), USING THE U() OPTIONS
14:17.06sibiria^
14:17.09Asteriscoohhh
14:17.20Samot9:48:13 AM <Samot> That is why Dial() has pre/post dial options to send calls to contexts for that
14:17.28sibiriayes, as said, Samot told you also minutes ago
14:17.35SamotIt was literally the second thing I said in all of this.
14:17.39Asteriscoexcuse me :)
14:17.40sibiriathe kludge here is that in your post-dial context you will need to inspect the channel
14:17.49sibiriastring comparison of the channel f.e.
14:18.15Asteriscocan you pastebin the right code?
14:18.16sibiriaor actually, you don't need that, since you will be using the post-dial _only_ in the IAX Dial()
14:18.32sibiriaAsterisco: wiki.asterisk.ork   search for Dial
14:18.38sibiriato find the documentation for Dial
14:18.47sibiriathe option is U, and it's described clearly there
14:18.53sibiriawiki.asterisk.org*
14:19.24sibiriaif you're using an older version of asterisk i think M() is the option you need
14:19.38sibiriaM() is now deprecated in favor of U() if i'm not mistaken
14:19.52SamotCorrect.
14:21.44Asteriscoexten => _0X,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)))
14:21.47Asteriscois right?
14:22.22sibiriano, U() wants the name of a context asterisk will jump to and execute
14:23.28Asterisco[chiamataanonima]
14:23.28AsteriscoPlayback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
14:23.37Asterisco...
14:23.37Asteriscoexten => _0X,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(chiamataanonima))
14:23.37sibiriaso Dial(SIP/123,,U(blah))       [blah] exten => s,1,Playback(beepboop)   etc.
14:23.42sibiriayes
14:23.50sibiriadon't forget to Return
14:23.55Asteriscoand then return to the same context?
14:24.01Asteriscohow?
14:24.02sibiriayes, just Return will be enough
14:24.19sibiriasame => n,Return
14:24.25Asteriscoahhh :)
14:24.28Asteriscook :
14:24.30Asterisco:)
14:24.34Asteriscoit's funny
14:24.40Asteriscothank you
14:24.50sibiriawait with thanking us until you've tested so it works
14:25.40Asterisco[chiamataanonima]
14:25.40Asteriscoexten => s, 1, Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
14:25.40Asteriscoexten => s, n,Return
14:25.43Asteriscoright?
14:25.57sibiriayep
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14:29.44Asteriscohttps://pastebin.com/yEqB3x6U
14:29.49Asteriscodon't work
14:30.01Asteriscohe try to use only firs line
14:30.55sibiriait will always use the first line
14:31.07sibiriabecause that's what your context does
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14:31.38Asterisco...if the first line is busy... it come to the second line
14:31.40Asteriscoor not?
14:31.58SamotRight but that doesn't show it moving the the next priority in the context.
14:32.18SamotIt matched on priority 1 and Dial() then hung up.
14:32.45Asterisco?!?!?!?!?
14:32.51SamotIs that what happened?
14:32.59SamotAsterisco: ?
14:33.10Asteriscoone moment
14:33.59Asteriscohttps://pastebin.com/Zx1kSyX6
14:34.05Asteriscowhere is the error?!?!?!?!
14:34.11SamotDUDE
14:34.14SamotDid you make a test call?
14:34.23Asteriscoi do it
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14:34.35SamotAnd it ONLY Dial'd the first line?
14:34.41Asterisconow i can call with 095448212 and stop
14:34.43SamotWhich came back BUSY, right?
14:35.00Asteriscoright
14:35.06SamotDid it dial the IAX trunk?
14:35.22Asteriscowhen i close the line start the second call that use always 095448212
14:35.42Asteriscoi cannot dial with iax trunk...
14:35.52Asterisco...is possible that is busy IAX
14:36.11Asteriscothis line is in another office
14:36.23Asteriscoand is possible that other people use it...
14:36.26Asterisco?
14:37.44SamotAsterisco: What is it that you want this to do?
14:37.56SamotDial the first trunk, then dial the second trunk?
14:37.57Asteriscouse iax line
14:38.01Asteriscoyes
14:38.09SamotBut it's not doing that is it?
14:38.18Asteriscothe first line is busy, use iax line
14:38.26Asteriscodon't work
14:38.28SamotOnly if it's BUSY?
14:38.42Asteriscoi can use first line 095448212...
14:38.48SamotStop.
14:38.51Asterisco...but the second call don't start
14:38.54SamotAnswer questions.
14:39.27SamotYou want to call the IAX trunk if the first trunk is BUSY?
14:39.48Asteriscoof course
14:39.56SamotBUSY is what way?
14:40.05SamotOut of channels? Or the destination was BUSY?
14:40.16Asteriscoout of channels
14:40.27SamotOK then you need to deal with the DIALSTATUS
14:40.48Asterisco??
14:40.51SamotAnd based on the proper DIALSTATUS then attempt to use the IAX trunk.
14:41.12AsteriscoSamot... but first if the line is busy
14:41.21Asteriscoasterisk try to use second line
14:41.40Asteriscoif second line was busy, asterisk try to use third line
14:41.46SamotFirst off, you don't want this on a BUSY
14:41.57SamotYou want this on a NO MORE CHANNELS
14:42.04SamotThat the provider will offer you.
14:42.19SamotThose are DIFFERENT responses from the provider.
14:42.43AsteriscoSamot can you correct my extensions? :\
14:42.59SamotNo because this is more than just three lines.
14:43.13Samotyou want to do things based on specific responses from the call.
14:43.20SamotYou're going to need to expand your dialplan.
14:43.29SamotAnd understand what you are doing with it.
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14:59.01AsteriscoSamot....
14:59.06AsteriscoHELP me! please
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15:13.17sibiriaso what is the problem then? i didn't follow the discussion
15:13.33sibiriait starts to dial the first time, and then what happens?
15:14.29Asterisconothing
15:14.44Asteriscodon't start the call using the second line
15:17.49sibiriawhen the first one is... busy?
15:17.58Asteriscoyes
15:18.07sibiriaDial() will continue in the dial plan if nothing is answered
15:18.08Asteriscowhen the first lin is used
15:18.20sibiriaare you using a hangup-handler or so?
15:18.22Asteriscono
15:18.29Asterisco?
15:18.36sibiriai would guess that means you are not using a hangup handler
15:18.36Asterisco<PROTECTED>
15:19.51AsteriscoSibiria---
15:19.52Asteriscohttps://pastebin.com/Zx1kSyX6
15:21.21sibiriahow do you know the first line is busy?
15:21.36SamotIts not busy
15:21.36sibiriai mean, how do you know what it signals
15:21.41SamotIts a 503
15:22.06sibiriaok i suppose that should still cause Dial() to just continue the dial plan execution
15:22.56Asterisco???
15:23.28SamotThe use of the word Busy here is for when the provider denies the call due to lack of capacity
15:24.12sibiriaAsterisco: can you show the log for the call?
15:24.27sibiriaand this isn't asterisk 1.8 i hope? :P
15:24.43Asteriscoi check
15:25.18AsteriscoAsterisk 13.14.1~dfsg-2+deb9u4, Copyright (C) 1999 - 2014, Digium, Inc. and others.
15:26.43sibiriaAsterisco: can you try adding option g to the first dial?
15:26.57sibiriadial(sip/blah,,g)
15:27.03sibiriajust to see what happens
15:27.42sibiria(don't forget to reload the dial plan first)
15:27.45Asteriscoexten => _0X,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(chiamataanonima)) ;
15:27.50Asteriscoexten => _0X,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,g) ;
15:27.55sibiriano not that dial
15:27.56sibiriathe FIRST one
15:27.58Asteriscook
15:28.04sibiriafor @095448212
15:28.24Asteriscoexten => _0X.,1,Dial(SIP/${EXTEN}@095448212,,g) ;
15:28.28Asteriscoright?
15:28.30sibiriayes
15:30.16Asteriscohttps://pastebin.com/RdKi9wyN
15:30.42sibiriathat call was answered
15:30.45sibirianot busy
15:30.50sibiriaor otherwise failed
15:31.19Asterisco:\
15:31.57Asteriscowhen i call 095505040 Asterisk use @095448212
15:32.00Asteriscoand is right
15:32.35Asteriscobut when i try to call 0957179627 Asterisk use always @095448212 and not the IAX line :\
15:33.12sibiriano matter what you call, it will always start on dial(...095448212)
15:33.18sibiriait is the first step in your context
15:33.21sibiriait will _always_ go there
15:34.09Asteriscoof course...
15:34.37Asteriscobut if first line is used from other.... Asterisk must use the second IAX line
15:34.39Asteriscoor not?
15:35.22sibiriaif Dial() can't connect the call, it will continue with the next priority
15:35.48sibiriaoption 'g' for Dial makes it do the same even if the call was answered. i asked you to test this just to see what happens
15:36.02sibiriabut to me it looks like your asterisk setup can't (or does not want to) dial with IAX
15:36.22Asteriscosibiria Iax works fine!!
15:37.00Asteriscobefore that i modify this: exten => _0X,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(chiamataanonima)) ;
15:37.05sibiriaso you are sure chan_iax2 is loaded?
15:37.11Asteriscoyes
15:37.22sibiriawhat if you comment out the other two dials and have _only_ the ${IAX_EUROPTICAL} dial?
15:37.31Asteriscoyes
15:37.37sibiriaso test it
15:37.41sibirialet's see it
15:37.49Asteriscoi retry
15:37.58Asteriscoexten => _0X.,n,Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
15:37.58Asterisco;poi proviamo ad usare la linea 095500052 per effettuare chiamate ai cellulari
15:37.58Asteriscoexten => _0X.,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30) ;Tutte le chiamate verso i numeri fissi dobbiamo aggiungere il prefisso *67#
15:38.04sibiriaand what does IAX_EUROPTICAL contain?
15:38.04Asteriscoexten => _0X,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(chiamataanonima)) ;
15:38.49AsteriscoIAX2/cerza:password@europtical
15:39.18SamotUhm, guys.
15:40.05SamotHow does that outgoing context get called?
15:40.27sibiriahe includes it i think
15:40.43sibiriai would personally just avoid leaning into it from another context *shrug*
15:40.47Asteriscosibiria is strange
15:40.55Asteriscoiax don't working!
15:41.01sibiriashow the log
15:41.24Asterisco<PROTECTED>
15:41.24Asterisco<PROTECTED>
15:41.24Asterisco<PROTECTED>
15:41.24Asterisco<PROTECTED>
15:41.24Asterisco<PROTECTED>
15:41.26Asterisco<PROTECTED>
15:41.40sibiriathat's not IAX
15:41.44sibiriathat is chan_sip
15:41.54SamotShow what is in the [interni] context.
15:42.23sibiriai get the feeling you're not even reaching [outgoing]
15:45.28Asterisco-- Called IAX2/cerza:xunil@europtical/*67#095430666
15:45.28Asterisco<PROTECTED>
15:45.28Asterisco<PROTECTED>
15:45.42SamotShow what is in the [interni] context.
15:45.44Asteriscoif i exclude the first Dial... Iax work fine
15:45.47SamotShow what is in the [interni] context.
15:45.54Asterisco[interni]
15:45.54Asteriscoinclude => numeri_interni
15:45.54Asteriscoinclude => alias
15:45.54Asteriscoinclude => outgoing
15:45.54Asteriscoinclude => fromiax
15:45.55Asteriscoinclude => incoming
15:46.34sibiriatry this and show the log output of the call: https://pastebin.com/raw/9q9xjxvK
15:46.53Asteriscook
15:46.58sibiriaremember:   dialplan reload
15:47.04sibiriaafter editing the changes
15:49.13Asterisco[Jun 28 17:48:38] WARNING[5485][C-00000000]: chan_sip.c:33735 process_crypto: Ignoring crypto attribute in SDP because RTP transport is insecure
15:49.13Asterisco<PROTECTED>
15:49.13Asterisco<PROTECTED>
15:49.13Asterisco<PROTECTED>
15:49.14Asterisco<PROTECTED>
15:49.16Asterisco<PROTECTED>
15:49.18Asterisco<PROTECTED>
15:49.20Asterisco<PROTECTED>
15:49.24Asterisco<PROTECTED>
15:49.26Asterisco[Jun 28 17:48:48] ERROR[5532][C-00000000]: app_stack.c:594 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:chiamataanonima, Extension:s, Priority:1)
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15:50.20sibiriaright so it works, besides the missing context playing the audio file
15:50.34AsteriscoNo
15:50.52Asteriscoi can't listen the audio :\
15:50.52Samotsibiria: He didn't adjust his dialplan, he replaced it.
15:50.58sibiriayes; your IAX2 peer works
15:51.00SamotMaking the IAX trunk the first priority.
15:51.19Asteriscosamot i do it!
15:51.30Asteriscoexten => _0X.,1,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(chiamataanonima))
15:51.30Asterisco<PROTECTED>
15:51.50SamotOK that's great an all
15:52.00SamotBut the IAX trunk wasn't even being dialed to being with in the other setup.
15:53.39Asterisco[Jun 28 17:53:11] ERROR[5555][C-00000001]: app_stack.c:594 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:chiamataanonima, Extension:s, Priority:1)
15:53.56Asteriscoah ok
15:54.01Asteriscoexten => s,Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
15:54.11Asteriscomust be exten => s,1,Playback(/web-asterisk/asterisk/audio/chiamata_anonima-8khz)
15:54.25sibiriait's not being included
15:54.31sibiriainclude => chiamataanonima
15:54.39Asteriscoahhh :)
15:55.19sibiriaor better yet, define it above [interni] actually
15:55.31Asterisco- IAX2/europtical-1171 Internal Gosub(chiamataanonima,s,1) start
15:55.31Asterisco[Jun 28 17:55:22] ERROR[5703][C-00000000]: app_stack.c:594 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:chiamataanonima, Extension:s, Priority:1
15:55.55Asterisco...
15:55.56sibiria[tomatoanonima]
15:55.56sibiria...
15:55.58Asteriscobut now...
15:55.58sibiria[interni]
15:55.58Asterisco[interni]
15:55.59Asteriscoinclude => numeri_interni
15:55.59Asteriscoinclude => alias
15:55.59Asteriscoinclude => outgoing
15:55.59sibiria...
15:55.59Asteriscoinclude => fromiax
15:56.01Asteriscoinclude => incoming
15:56.03Asteriscoinclude => chiamataanonima
15:56.06Asteriscoi've included it
15:56.09sibiria[tomatoanonima]
15:56.10sibiria...
15:56.12sibiria[interni]
15:56.12sibiria...
15:56.21sibiriaand remove include => anonimga
15:58.05Asterisco-- Executing [s@chiamataanonima:1] Playback("IAX2/europtical-16219", "/web-asterisk/asterisk/audio/chiamata_anonima-8khz") in new stack
15:58.05Asterisco<PROTECTED>
15:58.05Asterisco<PROTECTED>
15:58.05Asterisco[Jun 28 17:57:35] NOTICE[5850][C-00000000]: app_stack.c:1082 gosub_run: IAX2/europtical-16219 Abnormal 'Gosub(chiamataanonima,s,1)' exit.  Popping routine return locations.
15:58.26sibiriaok, good. your audio indication is now working
15:58.39Asteriscobut i can't listen it!!
15:58.55Asteriscodon't work yet
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16:00.51*** part/#asterisk mahafyi (~quassel@103.195.203.244)
16:00.56Asteriscois possible that the file is small
16:01.02Asteriscoonly 2 seconds
16:04.57sibiriachange sound file to en/hello-world
16:05.03sibiria(assuming you have this available)
16:05.06Asteriscoi do it
16:05.10Asteriscodon't work
16:05.26Asteriscoduring the play audio
16:05.32Asteriscoi can hear only ring
16:05.53Asteriscowhen it finsh, start the call
16:08.38SamotOK..
16:08.44SamotThe U() option is wrong.
16:08.44Asteriscook i find the problem
16:08.56Asteriscoin [chiamataanonima]
16:09.06Asteriscoi must use exten => _0X.,1,....
16:09.14Asteriscoand not exten => s,1,
16:09.31SamotIt's still wrong.
16:09.58Asteriscobut so i can hear the audio file
16:10.01SamotU() executes the GoSub on the CALLED channel after it is ANSWERED but before connected to the CALLING party
16:10.06SamotOK
16:10.20Asteriscoone moment
16:10.30sibiriaU() is for doing something _before_ bridging the call, to avoid "them" f.e. hearing the audio
16:10.40SamotOn the CALLED channel.
16:12.53Asteriscoafter playing audio... don't start the iax call
16:13.01Asterisco> 0x7f9f980038e0 -- Strict RTP learning complete - Locking on source address 10.0.1.51:5006
16:13.01Asterisco<PROTECTED>
16:13.01Asterisco[Jun 28 18:11:56] ERROR[6639][C-00000000]: app_stack.c:391 return_exec: Return without Gosub: stack is unallocated
16:13.01Asterisco<PROTECTED>
16:14.40sibiriawithout gosub.... you are using U()?
16:14.47sibiriaU() _should_ jump with a gosub
16:17.31Asteriscoexten => _0X.,1,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,U(chiamataanonima))
16:17.54Asteriscois really strange
16:18.09Asterisconow i dont' have any error
16:18.17Asteriscobut i can't hear the audio file
16:18.31Asteriscowile it "playing" i can see only ring
16:18.37SamotRight. Because U is the wrong option.
16:18.47sibiriayou could try B() to keep it in the original channel
16:18.53sibiriabut that would play even if the call fails
16:19.03Asterisco:\
16:20.01Asterisco[Jun 28 18:19:40] NOTICE[7092][C-00000000]: pbx.c:2917 pbx_extension_helper: No such label 'chiamataanonima' in extension '095430666' in context 'interni'
16:20.01Asterisco[Jun 28 18:19:40] WARNING[7092][C-00000000]: pbx.c:8597 pbx_parseable_goto: Priority 'chiamataanonima' must be a number > 0, or valid label
16:20.01Asterisco[Jun 28 18:19:40] ERROR[7092][C-00000000]: app_stack.c:575 gosub_exec: Gosub address is invalid: 'chiamataanonima'
16:20.54Asteriscoi've used B()
16:21.08sibiriaB(anonima^s^1)
16:21.18Asteriscook
16:21.32Asteriscoexten => _0X.,1,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,B(chiamataanonima^s^1))
16:22.03SamotIt's going to play back the recording before the call is even started.
16:22.17Asteriscook
16:22.24Asterisconow works fine :)
16:22.40SamotExcept it will play regardless if busy or not.
16:22.41sibiriaexcept that it will always play that audio file, even if no one picks up or if it's busy etc.
16:22.59sibiriamaybe there's a way to direct playback to a certain channel, i don't really know
16:23.26Asterisconow i try...
16:23.27Asteriscoexten => _0X.,1,Dial(SIP/${EXTEN}@095448212,,g) ;
16:23.27Asteriscoexten => _0X.,n,Dial(${IAX_EUROPTICAL}/*67#${EXTEN},30,B(chiamataanonima^s^1))
16:24.34AsteriscoFINEEEE
16:24.40AsteriscoNow works
16:24.42Asterisco:)
16:24.49SamotThat's going to call the IAX trunk if the call isn't answered.
16:24.54AsteriscoSamot is not important...
16:25.10SamotYou said that it should only do it when the SIP trunk has no more lines.
16:25.17SamotSo now that's not important?
16:25.22sibiriait's going to call it if it's answered, too :)
16:25.31Asterisco<Samot> Except it will play regardless if busy or not.
16:25.34sibiriayou should really not use 'g' here - i only wanted you to test it
16:25.35AsteriscoThis is not important
16:25.39SamotThe B() option.
16:25.44SamotI'm talking about the SIP trunk.
16:25.54Asteriscoi remove ,g
16:26.06SamotIf a call goes out the SIP and is not answered, you have nothing in the dialplan to deal with the proper dial status.
16:26.23SamotSo if no one picks up the SIP trunk, it's still going to call them again on the IAX trunk.
16:26.26sibiriai would also check ${DIALSTATUS} between each call
16:26.30SamotRight.
16:26.33sibiriaa conditional jump. much safer
16:26.48SamotBecause you only want a certain dialstatus to call the IAX trunk.
16:28.24Asteriscosibiria
16:28.34Asteriscohow can i check Dialstatus?
16:34.46sibiriaf.e. GotoIf($["${DIALSTATUS}" != "BUSY"]?ok)
16:35.11sibiriathen end that context with the label "ok" which just NoOps or whatever
16:35.36sibirialast in the context: same => n(ok),NoOp
16:36.13sibiriaoh != "ANSWER" obviously :)
16:36.36sibiria= "ANSWER"*
16:37.04sibiriasorry, trying to do multiple things at once here. brain is not working properly in this 28°C heat
16:38.12Asteriscothank you :)
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18:46.49_ganapathi_updated asterisk version to 16.4.0 . but after that wav format on moh not working.
18:46.55_ganapathi_what would be the reason ?.
18:47.14_ganapathi_is there any module need to be enabled or need to check apart from music on hold module?
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21:56.23Kobazwhat do you guys use for converting pdf to multi-page tiff for SendFax()
22:07.13Kobazweeeee
22:08.02Kobazokay, so first i need to clean up the pdf: pdftocairo -pdf /tmp/T38_TEST_PAGES.pdf /tmp/send_fax.pdf; and then i can: convert /tmp/send_fax.pdf /tmp/send_fax.tiff
22:08.07Kobazwow that was a pain to figure out
22:08.22Kobaz(if anyone is interested)
22:09.18Kobazthere was just something in the pdf that was causing convert to tiff to completely fail
22:10.42*** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca)
22:32.02Kobazand then the key ingrediant convert to a fax compatible tiff: gs -q -sDEVICE=tiffg3 -r204x196 -dBATCH -dPDFFitPage -dNOPAUSE -sOutputFile=out.tif in.pdf
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